[asterisk-users] Calls Being Randomly Bridged

Mon Jan 21 19:31:07 CST 2008

Wow thanks so much for this, this is a lot of great info.  Hopefully
enough to catch snom's attention to.  Is it possible for you to try 7.x
on one of the phones and see if it corrects the problem?

What it comes down to, is that the phone is too complicated to handle
multiple calls for non technical users.  They have to keep track of way
too much, even a techie like us could get mixed up sometimes, especially
in a high stress doctors office where there are half of the number of
receptionists that are reeally needed.

Mike

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve
Davies
Sent: Monday, January 21, 2008 9:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Calls Being Randomly Bridged

I found this problem sufficiently interesting that I went and had a play
with our snom phones in the test lab to try and determine what the
behavious is. This is with 6.5.13 phones, and I think the results are
somewhat inconsistent, particularly if snom are reporting this behaviour
as "intended" as was suggested elsewhere in this thread...

We already disable the "Call join on Xfer (2 calls):" setting, so that
can be taken into account in the descriptions below.

1) Simple unattended transfer. This does what is says on the tin
regardless of how many other calls are ringing one the handset. It will
transfer the call that is "in-hand" to the number dialled.

Achieved with: Transfer, dial number, Tick

2) Simple attended transfer - One caller on the line. Again, this works
fine

Achieved with: Hold, dial number, tick, wait for answer, transfer, tick
Or: Hold, dial number, tick, wait for answer, Hangup
Or: Hold, dial number, tick, wait for answer, Transfer, Tick

3) With multiple inbound calls, the behaviour is less well defined.
Here is what I found:

  Call 1 arrives, answer call.
  Call 2 arrives
  Call 3 arrives
  Press hold, dial destination for transfer of call 1, press Tick.

Now there are 2 alternatives.

a) Unattended. While the call is still ringing, press transfer, you will
be offered a list of calls in the order 1, 3, 2 - This is 100% fine. The
default destination is call 1 - The last call we dealt with.

b) Attended. Wait for the call to answer, Press transfer, you will be
ordered a list of calls in the order 3, 2, 1 - This is 100% wrong. The
call you want is LAST in the list. If you have no CID, or have forgotten
the CID of the caller, you cannot easily transfer the right call, and
might instead connect the wrong caller. Why would you offer an
unanswered call over an answered one anyway???

4) How to connect two external callers (as per original email). This is
a stretch, but I can see it happening...

Answer a call, put it on hold, wait for an answer. Re-select the
original caller's line to let them know you are about to transfer their
call. Press transfer (another call has come in in the meantime) the list
you are offered defaults to the new (unanswered) call, and not the
recently dialled and answered transferee.

Not good really :(

Basically, whatever calls the operator has had DIRECT involvement with
should be kept at the top of the "stack" of calls, so that any default
operations relate to those topmost calls. New calls go at the bottom of
the stack, and stay there until there is some direct interraction with
them. How hard is that?

Just my 2p.

Steve

> >
> > -----Original Message-----
> > Date: Sat, 19 Jan 2008 21:32:42 -0500
> > From: "Michael J. Liberatore" <mike240se at straightandnarrowinc.org>
> > Subject: [asterisk-users] Calls Being Randomly Bridged
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> >         <asterisk-users at lists.digium.com>
> >
> > Hi i have a friend who i setup an asterisk system for at his doctors

> > office.  it has 3 snom 360 phones with 6.2.x stable firmware and 
> > latest  asterisk 1.4 and zaptel.  They have the digium 4 port fxo
card.
> >
> > They are extremely upset because calls are being randomly bridged 
> > for no rhyme or reason.  They say that callers will call in and 
> > sometimes get  connected with other callers, or they will be in the 
> > queue and then be talking to another caller waiting in the queue or 
> > on hold.  Or they will be talking to a patient and then have another

> > patient end up on the  conversation.
> >
> > They are freaking out because of hippa and laws that govern privacy 
> > but i have no clue why.  I assume most cases are conference calls 
> > being initiated by accident.
> >
> > So any help would be greaat.  maybe just disabling conference calls 
> > would be a good start but i dont know how with sip phones.  or maybe

> > this is a bug?  unfortuinately they dont give me much info and i 
> > dont use the phones so i dont have any specific logs to show, they 
> > just call  me freaking out saying this stuff but they rarely can 
> > give me a specific call cause they get so many.
> >
> > thanks
> >
> > mike

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